FIG. 1 illustrates a network 100 capable of carrying voice, video, and multimedia traffic as well as traditional data. In a preferred embodiment, the network 100 uses the IP protocol as a network layer protocol, and uses a combination of protocols generally known as Voice over IP (VoIP) to carry the voice, video, and/or multimedia traffic over the network layer. Routers 110 forward traffic within network 100. A gateway 120 connects network 100 to the PSTN 130.
A communication endpoint 140 in communication with the network 100 can make and receive calls (voice, video, and/or multimedia) using the facilities of the network 100. A call includes multiple streams of VoIP packets traveling over the network 100: signaling packets to establish (set up), terminate (tear down), modify, and monitor a call; media packets to carry the voice, video, and/or multimedia content; and optional media control packets which carry statistics and timing information related to the call (e.g., jitter, latency, etc.). Various protocols can be used for each of these packet types, although the most common protocols are Session Initiation Protocol (SIP) for signaling packets, Real-time Transport Protocol (RTP) for media packets, and Real-time Transport Control Protocol (RTCP) for media control packets.
A conventional session border controller (SBC) 150 resides at the edge of network 100, and all signaling, media and media control packets for endpoint 140 pass through the SBC 150. SBC 150 provides session routing based on rules and policies, in which multiple signaling routes are examined. Various distribution strategies are used to select a particular signaling destination. SBC 150 also provides media routing so that network routers prioritize call traffic over data traffic. Without media routing and session routing, packets would flow whichever way the underlying network topology would allow, thereby subjecting multi-media data packets to disruptive paths, as well as upstream and downstream failures.